search for: sdps

Hi all, Sorry if this is out of place, but I've seen people suggesting future functionality for Dovecot on here before. TBH I'm not completely sure if this would be possible in a LDA. Demon (an ISP) have extensions to POP3 they call "SDPS" - there's basically an extra command that can be called for an email and it gives the envelope rcpt and from addresses. I was wondering if there was any chance of future versions of Dovecot allowing something similar? Many thanks, Mark Lidstone IT and Network Support Administrator BMT...

...e nephews can have their own lines. However, one of the phones I got was the SNOM 200. That's worked fine for me on my own network, but I'm having bad luck getting it to work behind NAT talking to Asterisk. It talks to my termination/origination provider, which seems to ruthlessly ignore SDPs and send audio to the address it gets audio from, which works pretty well behind NAT. I've tried all the various NAT settings on the SNOM 200 (with the last firmware rev they made) but reports are that's broken. The SDPs and Contact headers it sends out are always the natted address, even...

Hi List Asterisk 16.28.0 in use. PJSIP in use Two endpoints Both using IPv6 One Endpoint on UDP, the other via TLS. Both with: t38_udptl=yes ;fax_detect=yes ;fax_detect_timeout=30 rtp_ipv6=yes Both sides are T.38 capable and detect fax tone so no need for fax detection on asterisk. Voice calls between the two work fine. But on a Fax call, I see this situation: A <=> Asterisk

Hello, We are trying to send faxes by T.38 protocol to a remote SIP proxy from a local extension. The local extension sends the INVITE, Asterisk sends the call to the Proxy the call is connected with a regular audio codec. After a few seconds the remote proxy sends an INVITE with UDPTL and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just

...I completion on the endpoint (Reported by Alexei Gradinari) * ASTERISK-27980 - Caller ID cannot be changed on Attended Transfer before dialing out (Reported by Alexei Gradinari) * ASTERISK-28107 - app_confbridge: Participant info labels aren't being added to the SDPs (Reported by George Joseph) * ASTERISK-28089 - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Rep...

...I completion on the endpoint (Reported by Alexei Gradinari) * ASTERISK-27980 - Caller ID cannot be changed on Attended Transfer before dialing out (Reported by Alexei Gradinari) * ASTERISK-28107 - app_confbridge: Participant info labels aren't being added to the SDPs (Reported by George Joseph) * ASTERISK-28089 - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Rep...

Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070622/43308a1f/attachment.htm

Hello, Has any more work been done on this front? The message below is from Sept. 2006. In particular, the LT/TXT Technology Enabling Platform (TEP) is now available from MPC Corp. Where can one obtain an appropriate AC SINIT module (i.e., like lpg_sinit_20050831_pae.auth.bin below)? I would like to begin using Xen with TXT support. Thanks, -Jon This patch adds SMP support to the

...I completion on the endpoint (Reported by Alexei Gradinari) * ASTERISK-27980 - Caller ID cannot be changed on Attended Transfer before dialing out (Reported by Alexei Gradinari) * ASTERISK-28107 - app_confbridge: Participant info labels aren't being added to the SDPs (Reported by George Joseph) * ASTERISK-28089 - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) * ASTERISK-28076 - bridging: Asterisk crashes when receiving an empty realtime text frame (Rep...

...org/jira/browse/ASTERISK-27980>] - Caller ID cannot be changed on Attended Transfer before dialing out (Reported by Alexei Gradinari) - [ASTERISK-28107 <https://issues.asterisk.org/jira/browse/ASTERISK-28107>] - app_confbridge: Participant info labels aren't being added to the SDPs (Reported by George Joseph) - [ASTERISK-28089 <https://issues.asterisk.org/jira/browse/ASTERISK-28089>] - function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) - [ASTERISK-28076 <https://issues.asterisk.org/ji...